SIP trunking connects IP communications infrastructure to the PSTN using the Session Initiation Protocol, letting developers route inbound and outbound calls over broadband. This guide covers how SIP trunks work, authentication models, codec requirements, failure modes, and how Teler’s programmable voice API sits above the SIP layer for AI phone agent development in 2026.
SIP Trunking Explained for Developers Read More »
This guide covers how to build Indic-language voice agents using the Sarvam API for speech recognition and synthesis, paired with Teler for PSTN connectivity and real-time media streaming. Sarvam’s saaras:v3-realtime WebSocket endpoint accepts 16 kHz PCM audio, while telephony delivers 8 kHz narrowband audio, so resampling sits in every production path.
Building Indic-Language Voice Agents with Sarvam and Teler Read More »
Sarvam AI text-to-speech (Bulbul v3) generates natural Indian-language speech via REST and WebSocket, but it has no built-in PSTN or SIP layer. This guide covers the audio path, format requirements, and the WebSocket bridge you need to stream synthesized audio into a live phone call in 2026.
How to Put Sarvam AI Text-to-Speech on a Live Phone Call Read More »
A step-by-step guide on how to build an AI voice agent that makes real phone calls using the STT-to-LLM-to-TTS pipeline. Covers the 8 kHz μ-law audio format constraint, WebSocket media streaming, barge-in handling, and what you must build yourself versus what the telephony layer provides.
How to Build an AI Voice Agent That Makes Phone Calls Read More »
Choosing between WebSocket and SIP for AI voice agents in 2026 comes down to where your media terminates and how much telephony infrastructure you want to own. This guide covers the audio path, protocol trade-offs, failure modes, and the exact parameters that determine whether your agent sounds natural or choppy on a live call.
WebSocket vs SIP for AI Voice Agents Read More »
Voice agent latency determines whether a conversation feels natural or broken. This guide covers the full STT-LLM-TTS pipeline, Opus codec frame sizing, WebSocket transport, barge-in handling, and the specific Teler configuration parameters that keep end-to-end packet delivery under 100 milliseconds in 2026.
How to Hit Sub-100ms Latency in a Voice Agent Read More »
This guide covers LiveKit SIP outbound calling integrated with Teler’s WebSocket transport layer, verified against LiveKit v1.13.7 and Teler’s API as documented in September 2026. Learn how audio flows from carrier to model, which codecs matter, and what breaks in production when RTP media does not arrive on time.
LiveKit Outbound Calling over Teler SIP: Complete Guide Read More »
This guide compares the 9 best AI voice agent platforms available to US developers in 2026, evaluating each on real-time WebSocket latency, model-agnostic architecture, pricing transparency, and carrier compliance. FreJun Teler leads the list with low-latency targets, outbound calling from $0.20 per minute, and zero AI provider lock-in.
Best AI Voice Agent Platform: 9 Compared Read More »
Automated phone calls use a prerecorded or AI-generated voice to deliver messages without a live person dialing. This guide covers how they work, what they cost, the US compliance rules every business must follow in 2026, and when to keep a human on the line instead.
Automated Phone Calls: A Developer’s Guide Read More »
Choosing the best voice API for AI agents in the US means evaluating real-time WebSocket latency, model-agnostic architecture, and carrier compliance. This guide compares 9 providers, including FreJun Teler, Twilio, Telnyx, Plivo, Bandwidth, Sinch, Vonage, SignalWire, and Infobip, on pricing, integration paths, and AI readiness for 2026.
Best Voice API for AI Agents: 9 Options Compared Read More »
